[..] |
ASTERISK-30000: chan_dahdi: Add POLARITY function |
ASTERISK-30001: db: Removing nonexistent entries shows "Database entry removed" |
ASTERISK-30002: app_meetme: Don't erroneously set global variables when channel is NULL |
ASTERISK-30003: chan_dahdi: Allow fake ringing to be inhibited when immediate=yes |
ASTERISK-30004: chan_dahdi: Allow flash to hold to time out to silence |
ASTERISK-30005: res_pjsip: Crash when retrieving session due to destroyed dialog |
ASTERISK-30006: res_pjsip: UDP transport does not work when async_operations is greater than 1 |
ASTERISK-30007: chan_iax2: Prevent crashes due to attempted encryption with missing secrets |
ASTERISK-30008: samples: Remove obsolete config files |
ASTERISK-30009: Send provided DTMF to given Channel |
ASTERISK-30010: video message not sent via obd call in asterisk |
ASTERISK-30011: Testsuite: notify race in resource list subscription |
ASTERISK-30012: chan_iax2: register statement doesn't accept IPv6 address |
ASTERISK-30013: core_local: Local channels cannot have slashes in the destination |
ASTERISK-30014: Documentation on how to configure CISCO ASA 5505 firewall |
ASTERISK-30015: pjsip / WebRTC: Chrome creating large number of SDP attributes |
ASTERISK-30016: logger: rotate rotatestrategy replaces ${filename} with wrong name in exec_after_rotate |
ASTERISK-30017: Sip ! DialString to Pjsip |
ASTERISK-30018: app_meetme: MeetmeList AMI event not documented |
ASTERISK-30019: AMI OriginateResponse's Reason is NOT correct |
ASTERISK-30020: ConfbridgeListRooms Event Not Documented |
ASTERISK-30021: ast_variable_list_replace_variable uses variable with new keyword |
ASTERISK-30022: res_pjsip: Qualify is not updated with new Contact on refresh |
ASTERISK-30023: cdr_adaptive_odbc: does not support DATETIME database columns |
ASTERISK-30024: Failed to sign STIR/SHAKEN payload with functionality not enabled |
ASTERISK-30025: PJSIP affected by delays in DNS resolution |
ASTERISK-30026: ERROR[7334]: res_stir_shaken.c:1196 ast_stir_shaken_sign: Failed to retrieve certificate for caller ID |
ASTERISK-30027: ari: expose channel driver's unique id (i.e. Call-ID for chan_sip/chan_pjsip) in ARI channel resource |
ASTERISK-30028: Can't make outgoing calls : sip/2.0 401 unauthorized |
ASTERISK-30029: build: Git security vulnerability fix is sad with our accessing git as root during "make install" |
ASTERISK-30030: we are using asterisk 19 and when we are passing user =phone one time but in sip traces getting 2 times |
ASTERISK-30031: res_pjsip: User=phone being added multiple times |
ASTERISK-30032: Support of mediasec SIP headers and SDP attributes |
ASTERISK-30033: we are using asterisk 19 and when we are passing user =phone one time but in sip traces getting 2 times |
ASTERISK-30034: re_invite rejected with 491 after update of session_media with its SDP |
ASTERISK-30035: ari: bridge addChannel race condition causes segfault |
ASTERISK-30036: app_confbridge: Add CONFBRIDGE_CHANNELS function |
ASTERISK-30037: Add test support to calling external processes |
ASTERISK-30038: Asterisk Dynamically set outbound caller ID from mysql database |
ASTERISK-30039: cli: Targeted debug on startup deadlocks and creates unstable system |
ASTERISK-30040: Cant register to voip.ms |
ASTERISK-30041: Atxfer disconnects in 10 secs |
ASTERISK-30042: res_pjsip_transport_websocket: Registration over websocket returns a rewritten contact |
ASTERISK-30043: Wrong party is disconnected when hook-flashing on 3-way bridge |
ASTERISK-30044: GCC 12 issues |
ASTERISK-30045: Add test coverage to res/res_crypto.c functionality |
ASTERISK-30046: Reimplement res/res_crypto.c internals with EVP_PKEY interface to Openssl API's |
ASTERISK-30047: Local IP address in SDP when external_media_address is used |
ASTERISK-30048: How to add SDP in INVITE for pjsip |
ASTERISK-30049: Seeing unit-test failures in unadulterated master in /main/message |
ASTERISK-30050: Upgrade Asterisk to bundled pjproject 2.12.1 |
ASTERISK-30051: res_pjsip: No video after un-hold with moh_passthrough=yes |
ASTERISK-30052: add configuration setting to enable RTCP PLI, FIR and NACK independently from "webrtc" setting |
ASTERISK-30053: CHANGE SIP OPTIONS a= |
ASTERISK-30054: Asterisk => .Net Integration |
ASTERISK-30055: res_rtp_asterisk: HIgh CPU load when TURN server unreachable |
ASTERISK-30056: res_crypto.c build warnings |
ASTERISK-30057: channels: assertion in ast_waitfor_nandfds |
ASTERISK-30058: Evaluate dialplan functions and variables in agi exec |
ASTERISK-30059: menuselect: libxml include fails under Gentoo |
ASTERISK-30060: loader: format warnings in dev mode |
ASTERISK-30061: pbx: Add pbx helper application |
ASTERISK-30062: cli: Add CLI command to execute a dialplan app |
ASTERISK-30063: app_voicemail: Add option to prevent deletion of messages |
ASTERISK-30064: pbx: iax2 switch causes crash due to deadlock and assertion |
ASTERISK-30065: pjsip: Open Websocket connection is not reused for outgoing requests |
ASTERISK-30066: func_query: Add function to remotely query inter-Asterisk data |
ASTERISK-30067: Rewrite res_rtp_asterisk.c to use current API |
ASTERISK-30068: menuselect configure falure |
ASTERISK-30069: res_pjsip: Asterisk keeps qualifying after contact expiry |
ASTERISK-30070: pjsip: Deadlock between bridge, channel, and RTP |
ASTERISK-30071: rtp: Usage of rtp_timeout on WebRTC causes failure |
ASTERISK-30072: res_pjsip: allow TLS verification of wildcard cert-bearing servers |
ASTERISK-30073: Hello, Cisco 7821 telephone can register with Asterisk but we can not do call transfer. Could you please provide help for the correct configuration Thanks, Have a nice day |
ASTERISK-30074: Error while install asterisk 18.12.1 on Debian 10.12.0 |
ASTERISK-30075: say: Abort if channel hangs up during playback |
ASTERISK-30076: app_stack: Incorrect exit location in predial handlers logged |
ASTERISK-30077: Methods to initiate an Outbound call |
ASTERISK-30078: ari / pjsip: Push "identify" with hostname "match", fails due to invalid SQL query generation |
ASTERISK-30079: Randomly extension gets unregistered |
ASTERISK-30080: Call Dropping Issue - ASTERISK 11.5.1 |
ASTERISK-30081: app_confbridge: Channel can join wrong bridge due to race condition |
ASTERISK-30082: Getting this error while installing Asterisk 16 on Debian 9 |
ASTERISK-30083: chan_iax2: Optional dependency on openssl/res_crypto is now mandatory |
ASTERISK-30084: Agent status in queue is wrong |
ASTERISK-30085: res_parking: Dynamic creation takes a long time with large number of extensions |
ASTERISK-30086: res_parking: Warn when invalid parking space requested |
ASTERISK-30087: res_parking: Add music on hold override option |
ASTERISK-30088: func_curl: Subsequent calls are failing, can't clear headers |
ASTERISK-30089: general: fix typos |
ASTERISK-30090: xmldocs: Use example tags for examples |
ASTERISK-30091: cdr: Allow CDRs to ignore call state changes |
ASTERISK-30092: DateTime application: wrong inflection for one o'clock in German |
ASTERISK-30093: res_pjsip_refer: Pickup event is not sent from refer_incoming_invite_request |
ASTERISK-30094: app_queue: Periodically wrapuptime (pause after conversation) in application queue ends earlier than set in the configuration file queue.conf |
ASTERISK-30095: Unable to configure third-party/pjproject |
ASTERISK-30096: cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is unsupported at this time |
ASTERISK-30097: console: Recent documentation changes for connecting to remote console are inconsistent |
ASTERISK-30098: res_agi: DTMF input does not exit loop once I enter max input |
ASTERISK-30099: test_aeap_transport: transport_connect_fail sporadically causes failure |
ASTERISK-30100: res_pjsip: Path is ignored on INVITE to endpoint |
ASTERISK-30101: res_prometheus: Optional load res_pjsip_outbound_registration.so |
ASTERISK-30102: Hangup_handler not invoked |
ASTERISK-30103: chan_ooh323 Vulnerability in calling/called party IE |
ASTERISK-30104: Can't compile Dadhi using kernel 5.10.x |
ASTERISK-30106: res_calendar_icalendar: Microsoft online ICS calendars no longer work |
ASTERISK-30107: iostream: Build failure with libressl |
ASTERISK-30108: zaptel compile error which says "cannot find autoconf.h" |
ASTERISK-30109: res_pjsip: no contact-status AMI event on register of prune-on-boot contact that uses the same URI as before Asterisk restart |
ASTERISK-30110: chan_pjsip: endpoints that register elsewhere are not called from app_queue |
ASTERISK-30111: Segfault when a channel is hung up immediately after being bridged |
ASTERISK-30112: ari: Call recording stops prematurely |
ASTERISK-30113: ari: Loading PJSIP config wizard causes duplicated Endpoints, AORS, Auths created by push |
ASTERISK-30114: Real XSS on 8089? |
ASTERISK-30115: app_dial: Allow hook flashes to propogate on outbound dials |
ASTERISK-30116: Asterisk cel DB table issues |
ASTERISK-30117: pbx_lua: Remove compiler warnings |
ASTERISK-30118: Cant Install Asterik on (L)Ubuntu 22.04 LTS |
ASTERISK-30119: res_pjsip_sdp_rtp: Timeout doesn't handle bundled streams |
ASTERISK-30120: RTP Timestamp issue causes clicks & pops on calls routed via alternate Call Platforms |
ASTERISK-30121: Error In chan_iax2 |
ASTERISK-30122: GCC error when compiling Asterisk 18.13.0 with developer tools support |
ASTERISK-30123: features: Update automixmon documentation to reflect reality |
ASTERISK-30124: res_pjsip_outbound_registration: reload causes assertion in devmode and DO_CRASH |
ASTERISK-30125: wiki: Add AMI library documentation |
ASTERISK-30126: Spelling mistake in configs/samples/queues.conf.sample |
ASTERISK-30127: Create core Geolocation capability for Asterisk |
ASTERISK-30128: Create PJSIP interface module for Geolocation |
ASTERISK-30129: Use pre-dial and post-dial (bridge) handlers together in Dial command only executes the pre-dial handler |
ASTERISK-30130: app_queue: AMI "Event: AgentComplete" is sometimes missing data when channel is hung up by AMI |
ASTERISK-30131: Create Wiki documentation for Geolocation |
ASTERISK-30132: rtcp stats only for incoming channel |
ASTERISK-30133: no RTCP stats for outgoing channel |
ASTERISK-30134: IAX2 hints always return Unavailable |
ASTERISK-30135: [res_musiconhold] Allows the moh only for the answered call |
ASTERISK-30136: db: Add AMI action to retrieve all keys beginning with a prefix |
ASTERISK-30137: manager: Global disabled event filtered is incomplete |
ASTERISK-30138: Compile failure in res_geolocation/geoloc_eprofile.c when optimization is enabled |
ASTERISK-30139: Asterisk chan_iax2 error |
ASTERISK-30140: Asterisk chan_iax2 error |
ASTERISK-30141: remote_uri tests failing |
ASTERISK-30142: Missing port in Contact header in response to incoming requests |
ASTERISK-30143: manager: Read and Write output from "manager show connected" is not well documented/useful |
ASTERISK-30144: PRINTER CONNECTION PROBLEM |
ASTERISK-30145: GET FULL VARIABLE CHANNEL not working |
ASTERISK-30146: res_pjsip_logger: Add method-based log filtering |
ASTERISK-30147: Redis Dialplan functions and Device State |
ASTERISK-30148: 18.12.1 Crash / Sigabrt On Pjsip Transaction Destroy Under Load |
ASTERISK-30149: core: minmemfree watermark issue |
ASTERISK-30150: res_pjsip_session: Add support for custom parameters |
ASTERISK-30151: Documentation doesn't include info about "field", a 3rd required parameter. |
ASTERISK-30152: Documentation doesn't include info about "field", a 3rd required parameter. |
ASTERISK-30153: logger: Improve log levels |
ASTERISK-30154: pjsip: assertion failure |
ASTERISK-30155: core: CLI using -rx of "core show channels" returns empty |
ASTERISK-30156: sip_rtp_read: crash for video rtp session variable (vrtp) not inizialized |
ASTERISK-30157: chan_iax2: Deadlock with device state and channel locking |
ASTERISK-30158: PJSIP: Add new 100rel option "peer_supported" |
ASTERISK-30159: general: Remove obsolete SVN references |
ASTERISK-30160: cdr.conf: Remove obsolete app_mysql reference |
ASTERISK-30161: locks: add AMI event for deadlock |
ASTERISK-30162: when chan_iax is used to relay calls, no ringing indication is played |
ASTERISK-30163: general: fix minor formatting issues |
ASTERISK-30164: chan_iax2: Add missing option documentation |
ASTERISK-30165: Add AMI actions and events for Geolocation |
ASTERISK-30166: Add ARI objects and events for Geolocation |
ASTERISK-30167: res_geolocation: Refactor for issues found by users |
ASTERISK-30168: Asterisk GUI interface configuration |
ASTERISK-30169: res_pjsip: assertion on send_msg |
ASTERISK-30170: thirdparty: Certificate for svn.digium.com HTTPS is untrusted on Debian |
ASTERISK-30171: asttest (lua dependencies) / sipp: Compiler warnings |
ASTERISK-30172: Web socket closed abruptly |
ASTERISK-30173: Installation asterisk after menu select error messages chan_iax2.c |
ASTERISK-30174: RTP stream does not open between peers |
ASTERISK-30175: app_confbridge: Channels in ConfbridgeWelcome contains wrong muted information |
ASTERISK-30176: manager: GetConfig can read files outside of Asterisk |
ASTERISK-30177: res_geolocation: Add option to suppress empty elements |
ASTERISK-30178: extend user_eq_phone behavior to local uri's |
ASTERISK-30179: app_amd: Allow audio to be played while AMD is running |
ASTERISK-30180: app_broadcast: Add a channel audio multicasting application |
ASTERISK-30181: HTTP AMI sessions doesn't get purged fast enough (or at all) |
ASTERISK-30182: res_geolocation: Add built-in profiles to use in fully dynamic configurations |
ASTERISK-30183: SegFault / TCP Stack |
ASTERISK-30184: res_pjsip_session: re-INVITE after answering results in wrong stream direction of first call leg |
ASTERISK-30185: res_geolocation: Allow location parameters to be specified in profiles |
ASTERISK-30186: res_pjsip: Add support for reloading TLS certificate and key information |
ASTERISK-30187: chan_sip: Unsupported URI(416) |
ASTERISK-30188: add condition in chan_sip.c |
ASTERISK-30189: chan_pjsip: rtptimeout doesn't work at all when using Stasis Application |
ASTERISK-30190: res_geolocation: GEOLOC_PROFILE isn't returning correct values on incoming channel |
ASTERISK-30191: core/ari/pjsip: Add refer mechanism to refer endpoints to some resource |
ASTERISK-30192: res_tonedetect: fix typo for frametype |
ASTERISK-30193: chan_pjsip should return all codecs on a re-INVITE without SDP |
ASTERISK-30194: Jordan |
ASTERISK-30195: Dynamic sip id allocation to users |
ASTERISK-30196: dialplan for autocreatepeer=yes |
ASTERISK-30197: PauseMonitor hangs up the call |
ASTERISK-30198: Error `Too many open files` occurs after about ~8000 calls when using mixmonitor |
ASTERISK-30199: local direct media tests unstable |
ASTERISK-30200: res_pjsip: Reloading with removal of endpoint does not remove internal endpoint |
ASTERISK-30201: app_senddtmf: Repeat uses of PlayDTMF AMI action can cause infinite DTMF, or potential for crash |
ASTERISK-30202: install_prereq: don't require sudo |
ASTERISK-30203: Testsuite: apps/say_interrupt hangs forever |
ASTERISK-30204: res_pjsip_callerid: PJSIP and CALLERID(pres) / from_domain interaction |
ASTERISK-30205: testsuite: add missing pre-req |
ASTERISK-30206: runInVenv: Don't redirect output to /dev/null |
ASTERISK-30207: runtests: Use python3 in shebang |
ASTERISK-30208: runtests.py hangs forever if Asterisk cannot start |
ASTERISK-30209: pbx_variables: Use const char for pbx_substitute_variables_helper_full_location |
ASTERISK-30210: func_frame_trace: Channel masquerade triggers assertion |
ASTERISK-30211: app_confbridge: Add end_marked_any option |
ASTERISK-30212: Getting random segfault in libasteriskpj.so.2 |
ASTERISK-30213: Make crypto_load() reentrant and handle symlinks correctly |
ASTERISK-30214: mailbox_count_changes test failing intermittently |
ASTERISK-30215: Inbound SIP INVITE with Geo Location causing a Segmentation Fault |
ASTERISK-30216: app_bridgewait: Add option for BridgeWait to not answer |
ASTERISK-30217: Registration do not allow multiple proxies |
ASTERISK-30218: testsuite: Add Arch support to install_prereq |
ASTERISK-30219: Rejecting call from '0708060504' |
ASTERISK-30220: func_scramble: Fix segfault due to null pointer deref |
ASTERISK-30221: Channel masquerades screw up groups |
ASTERISK-30222: func_strings: Add trim functions |
ASTERISK-30223: features: add no-answer option to Bridge application |
ASTERISK-30224: Outbound proxy 'Route:' doubled in OPTIONS |
ASTERISK-30225: dahdi does not apply. |
ASTERISK-30226: REGRESSION: res_crypto complains about the stir_shaken directory in /var/lib/asterisk/keys |
ASTERISK-30227: Audiosocket: provision to exit cleanly from app_audiosocket |
ASTERISK-30228: Lot of „Autodestruct on dialog with owner SIP/XXXX in place (Method: BYE). Rescheduling destruction for 10000 ms“, and finally „taskprocessor.c: The 'stasis/p:channel:all' task processor queue reached 500 scheduled tasks“ |
ASTERISK-30229: PJSIP Configuration Wizard documentation should include mention necessary sorcery configurations |
ASTERISK-30230: Asterisk realtime database |
ASTERISK-30231: add answer delay to allow_sending_180_after_183 tests |
ASTERISK-30232: Initialize stack-based ast_test_capture structures correctly |
ASTERISK-30233: Add DAHDIBusyOut |
ASTERISK-30234: res_geolocation: ...may be used uninitialized error in geoloc_config.c |
ASTERISK-30235: res_crypto and tests: Memory issues and and uninitialized variable error |
ASTERISK-30236: app_queue.c: Properly escape commas in MONITOR* variables. |
ASTERISK-30237: res_prometheus: Crash when scraping bridges |
ASTERISK-30238: Got SIP response 503 “Service Unavailable” when i call |
ASTERISK-30239: Prometheus plugin crashes Asterisk when using local channel |
ASTERISK-30240: app voicemail odbc build error with gcc 11.1 |
ASTERISK-30241: res_pjsip_gelocation: Downgrade some NOTICE scope trace debugs to DEBUG level |
ASTERISK-30242: Persist queue member on pause after asterisk restart |
ASTERISK-30243: func_logic: IF function complains if both branches are empty |
ASTERISK-30244: res_pjsip_pubsub: Occasional crash when TCP/TLS connection terminated and subscription persistence is removed |
ASTERISK-30245: db: ListItems is incorrect |
ASTERISK-30246: pjsip: Allow topology/session refreshes in early media state |
ASTERISK-30247: Mistake in "Answer, Playback, and Hangup Applications" wiki documentation |
ASTERISK-30248: ast_get_digit_str adds bogus initial delimiter if first character not to be spoken |
ASTERISK-30249: Asterisk segfault - libcrypto.so |
ASTERISK-30250: remote party ID |
ASTERISK-30251: IAX2 Trunk Quality |
ASTERISK-30252: Unidirectional snoop on resampled channel causes garbled audio |
ASTERISK-30253: Regarding Setup of Latest Version of asterisk calling software on cloud |
ASTERISK-30254: res_tonedetect: Add audible ringback detection to TONE_DETECT |
ASTERISK-30255: prereqs: Missing test suite prereqs |
ASTERISK-30256: chan_dahdi: Fix format truncation warnings |
ASTERISK-30257: res_pjsip: IP addresses get butchered |
ASTERISK-30258: Dialing API: Cancel a running async thread, does not always cancel all calls |
ASTERISK-30259: ari: Crash on missing JSON validation in push registration |
ASTERISK-30260: ari: Creation of outbound PJSIP registration fails |
ASTERISK-30261: Video recording cannot play with application Playback. How can i play the recorded video |
ASTERISK-30262: res_pjsip_session: Allow a context to be specified for overlap dialing |
ASTERISK-30263: res_pjsip_notify: Allow using pjsip_notify.conf from AMI |
ASTERISK-30264: res_pjsip: Subscription handlers do not get cleanly unregistered, causing crash |
ASTERISK-30265: res_pjsip_session: Fix missing PLAR support on INVITEs |
ASTERISK-30266: Voicemail aborts when playing caller ID that includes + before digits |
ASTERISK-30267: ooh323 plugin is out of date with Asterisk |
ASTERISK-30268: After Channel move from Holding to Mixing sometimes audio is not heard between channels |
ASTERISK-30269: res_srtp: One way audio after unhold |
ASTERISK-30270: testsuite: Find/create deterministic user agent for attended transfer testsuite scenarios |
ASTERISK-30271: testsuite: Add yaml support to scenario iterator and migrate batch tests to use it |
ASTERISK-30272: webrtc client IP addresses behind reverse proxy |
ASTERISK-30273: test_mwi: compilation fails on 32-bit Debian |
ASTERISK-30274: chan_dahdi: Unavailable channels are BUSY |
ASTERISK-30275: ari: RTP packets being sent out of P2P with out of stream seqno/ts with early bridging |
ASTERISK-30276: res_pjsip: Mediasec requires different headers on 401 response |
ASTERISK-30277: Feature Transfer |
ASTERISK-30278: tcptls: Abort occurs if SSL error is logged if MALLOC_DEBUG is enabled |
ASTERISK-30279: Makefile: phoneprov sample files fail to install the first time |
ASTERISK-30280: Create capability to assign a Media Experience Score to RTP streams |
ASTERISK-30281: chan_rtp: Local address being used before being set |
ASTERISK-30282: CI: Coredump output isn't saved when running unittests |
ASTERISK-30283: app_voicemail: Fix msg_create_from_file not sending email to user |
ASTERISK-30284: app_mixmonitor: Add option to delete recording file when done |
ASTERISK-30285: manager.c: Remove outdated documentation |
ASTERISK-30286: app_mixmonitor: Add option to use real Caller ID for Caller ID |
ASTERISK-30287: webrtc |
ASTERISK-30288: Implement option to override User-Agent-Header on a per-registration basis |
ASTERISK-30289: xmldoc: Allow XML docs to be reloaded |
ASTERISK-30290: file.c: Don't emit warnings on winks. |
ASTERISK-30291: chan_sip: remove tests from testsuite |
ASTERISK-30292: chan_sip: convert chan_sip tests to pj_sip |
ASTERISK-30293: Memory leak in JSON_DECODE |
ASTERISK-30294: Many "task processor queue reached 500 scheduled tasks again" messages might cause Asterisk down. |
ASTERISK-30295: test_json: Remove duplicated static function |
ASTERISK-30296: Undefined symbol on naive RockyLinux 8 system |
ASTERISK-30297: chan_sip: Remove deprecated module |
ASTERISK-30298: chan_alsa: Remove deprecated module |
ASTERISK-30299: chan_mgcp: Remove deprecated module |
ASTERISK-30300: chan_skinny: Remove deprecated module |
ASTERISK-30301: res_pktcops: Remove deprecated module |
ASTERISK-30302: app_osplookup: Remove deprecated module |
ASTERISK-30303: res_monitor: Remove deprecated module |
ASTERISK-30304: app_macro: Remove deprecated module |
ASTERISK-30305: chan_dahdi: Allow FXO channels to start immediately |
ASTERISK-30306: require_odbc: Realtime table queue_log requires column 'time', but that column does not exist! |
ASTERISK-30307: unreal: Local Channel Not Optimizing |
ASTERISK-30308: pbx_builtins: Allow Answer to return immediately |
ASTERISK-30309: app_sla: Migrate SLA applications from app_meetme |
ASTERISK-30310: Sip Trunk with outbound proxy |
ASTERISK-30311: func_presencestate: Fix invalid memory access. |
ASTERISK-30312: Not able to Call |
ASTERISK-30313: threadpool: Control taskprocessor is blocked waiting for idle thread to terminate |
ASTERISK-30314: res_agi: RECORD FILE doesn't respect "transmit_silence" asterisk.conf option |
ASTERISK-30315: rtp port not accessible using kubeadm pods service ExternalIPs |
ASTERISK-30316: res_pjsip: Documentation should point out default if contact_user is not being set for outbound registrations |
ASTERISK-30317: Adding CodeSpaces config for the GitHub Mirror |
ASTERISK-30318: PINCHING CONNECTIONS |
ASTERISK-30319: Add BYE Reason support for SIP |
ASTERISK-30320: channel.c: Masquerades do not preserve tone playback |
ASTERISK-30321: Build: Embedded blobs have executable stacks |
ASTERISK-30322: res_hep: Add capture agent name support |
ASTERISK-30323: Calling ultiple Extensions |
ASTERISK-30324: codec_ilbc: Not compatible with ilbc 3.x.x |
ASTERISK-30325: Upgrade Asterisk to bundled pjproject 2.13 |
ASTERISK-30326: app_followme: Setting enable_callee_prompt=no(false) in followme.conf breaks timeout for calling from FollowMe application |
ASTERISK-30327: rtp_engine.h: Remove obsolete example usage |
ASTERISK-30328: Typo in from_domain description on res_pjsip configuration documentation |
ASTERISK-30329: app_dial: Option g() do not take effect while announce played |
ASTERISK-30330: callerid: Allow timezone to be specified at runtime |
ASTERISK-30331: sig_analog: Add full Caller ID support for incoming calls |
ASTERISK-30332: func_callerid: Warn if invalid redirecting reason provided |
ASTERISK-30333: chan_dahdi: Fix broken presentation for FXO caller ID |
ASTERISK-30334: res_pjsip: ca_list_path directive in pjsip.conf doesn't verify against all certificates |
ASTERISK-30335: pbx_builtins: Remove deprecated and defunct applications and options |
ASTERISK-30336: sig_analog: Fix no timeout duration |
ASTERISK-30337: res_pjsip_sdp_rtp: RTP not read before negotiation completes |
ASTERISK-30338: pjproject: Backport security fixes from 2.13 |
ASTERISK-30339: Prevent duplicate Asterisk processes from starting |
ASTERISK-30340: res_media_cache curl options configureable |
ASTERISK-30341: cdr_adaptive_odbc: fails to write CDRs after database reload |
ASTERISK-30342: Receive OPTIONS message, sometimes cannot send response "Unable to send response (-1)" |
ASTERISK-30343: res_crypto: ast_sign_bin fails with default RedHat crypto policies |
ASTERISK-30344: ari: Memory leak in create when specifying JSON |
ASTERISK-30345: loader.c: Modules that decline to load cannot be reloaded |
ASTERISK-30346: Fix NULL dereferencing issue in Geolocation |
ASTERISK-30347: xmldocs: Remove references to removed applications |
ASTERISK-30348: sig_analog: hidecallerid setting is broken |
ASTERISK-30349: app_if: Format truncation error |
ASTERISK-30350: res_pjsip_sdp_rtp: rtp_timeout_hold is not used when moh_passthrough has call on hold |
ASTERISK-30351: manager: Originate variables are not added when setvar used in manager.conf |
ASTERISK-30352: pbx_lua: Set callerid number when attended transfer |
ASTERISK-30353: func_frame_trace: Print text for text frames |
ASTERISK-30354: chan_iax2: Lack of formats prior to receiving voice frames causes jitterbuffer to stall |
ASTERISK-30355: ASTERISK-30248 Issue has come back in Asterisk 18.15.1 hangs up with trying to play + in the callerid on voicemail. |
ASTERISK-30356: suspect ps_endpints option rewrite_contact not rewriting with source ip. |
ASTERISK-30357: chan_dahdi: Allow automatic reoriginate on hangup |
ASTERISK-30358: OPUS doesn't generate SDP according to codecs.conf |
ASTERISK-30359: Install Prereq Script Enhancements |
ASTERISK-30360: utils: Refactor CURLOPT_WRITEFUNCTION callback |
ASTERISK-30361: json.h: Add missing ast_json_object_real_get |
ASTERISK-30362: res_pjsip_session: SDP o= header address changed in re-INVITE message |
ASTERISK-30363: Crash during srtp session |
ASTERISK-30364: IVR inputs |
ASTERISK-30365: Change voicemail filename |
ASTERISK-30366: app_queue: Fix preserve reason of pause when Asterisk is restared for realtime queues |
ASTERISK-30367: pbx: Fix outdated channel snapshots with pbx_exec |
ASTERISK-30368: app_queue: Opt-out calls are not tracked |
ASTERISK-30369: res_pjsip: Websockets from same IP shut down when they shouldn't be |
ASTERISK-30370: config: Template inheritance is incorrect for ast_variable_retrieve |
ASTERISK-30371: app_cdr: Remove deprecated NoCDR application |
ASTERISK-30372: sig_analog: Add Called Subscriber Held capability |
ASTERISK-30373: sig_analog: Add Call Waiting Deluxe options |
ASTERISK-30374: chan_dahdi: Allow automatic ADSI script downloads |
ASTERISK-30375: res_http_media_cache: Crash when URL has no path component. |
ASTERISK-30376: Disconnecting WebRTC SIP client disconnects all other WebRTC SIP clients |
ASTERISK-30377: Receiver and Caller can not hear the voice of each other |
ASTERISK-30378: Automatically removing all WebRTC Enpoints when closing one WebRTC Enpoint |
ASTERISK-30379: http: fix NULL pointer dereference while enable_status on TLS-only |
ASTERISK-30380: res_pjsip: Crash in simple |
ASTERISK-30381: res_resolver_unbound: Using unbound, queries do not try all available nameservers, and contacts will flap |
ASTERISK-30382: spam problems |
ASTERISK-30383: Asterisk crashes on snoop channel handle |
ASTERISK-30384: MixMonitorStart event is fired, but MixMonitorStop event is NOT fired |
ASTERISK-30385: Making a call from one extension(100) to another extension(200) through ari-client (Node.js) |
ASTERISK-30386: app_dial: Dial application has no escape character for option A |
ASTERISK-30387: res_pjsip_dialog_info_body_generator: Does not support call pickup |
ASTERISK-30388: res_phoneprov: Stale SERVER variable when multi-homed |
ASTERISK-30389: Asterisk version 13.30 show the error dahdi ------ Span 1: Channel 0/5 got hangup, cause 18 |
ASTERISK-30390: Asterisk version 18.15 show the error dahdi ------ Span 1: Channel 0/5 got hangup, cause 18 |
ASTERISK-30391: res_rtp_asterisk: Issue with transcoding g722 after MES changes |
ASTERISK-30392: Asterisk Log's time changing |
ASTERISK-30393: I'm trying to update the From header using dialplan , i'm setting it before dialing but its adding a new From header not updating the old one |
ASTERISK-30394: [patch] DUNDi process_clearcache() can result in deadlock |
ASTERISK-30395: install_prereq: sipp 3.5.2 compilation fails |
ASTERISK-30396: res_pjsip: Retrieval of contacts with + at start is problematic. |
ASTERISK-30397: SIP Reason: "Call completed elsewhere" no longer propagating |
ASTERISK-30398: Asterisk 20.1.0 missing application app_while |
ASTERISK-30399: call remote client connected to asterisk server |
ASTERISK-30400: Asterisk 16.28.0 Wireguard |
ASTERISK-30401: res_pjsip: Addition of endpoint/AOR requires notification for qualify/contact status |
ASTERISK-30402: app_queue: QueuePause generates an excessively large number of database queries |
ASTERISK-30403: VoiceMailMain. Problem with pronouncing CallerID Number. |
ASTERISK-30404: app_directory: Add reading directory configuration from custom file |
ASTERISK-30405: app_directory: Add 's' option to skip channel call |
ASTERISK-30406: pbx_ael: Global variables are not expanded. |
ASTERISK-30407: res_stir_shaken: Ordering of JSON fields incorrect, and tn lacks canonicalization |
ASTERISK-30408: Conflicting filenames in media cache cause wrong prompts to play |
ASTERISK-30409: Configurations before and after internet modem, for installation of asterisk and the set of hardware for telephone exchange |
ASTERISK-30410: Queues doesn't load members according to the configuration files |
ASTERISK-30411: app_read: add option to include terminating digit on empty, terminated strings |
ASTERISK-30412: chan_unistim: RTP Bleedover |
ASTERISK-30413: app_queue: UnpauseQueueMember generate generating string invalid in lastapp and lastdata |
ASTERISK-30414: features: attended 3 way transfer using the option atxferthreeway doesn't work as expected without atxferswap |
ASTERISK-30415: Asterisk DID range numbers cant able to show up |
ASTERISK-30416: Error sending STUN request: Invalid argument |
ASTERISK-30417: Copy/Paste error in UnpauseQueueMember |
ASTERISK-30418: Crash in ConfBridge |
ASTERISK-30419: pjsip: Crash when sending NOTIFY in PJSIP 2.13 |
ASTERISK-30420: res_speech_aeap: Crash due to NULL format on setup |
ASTERISK-30421: Callback too miscalls autodial |
ASTERISK-30422: app_senddtmf: add the option for senddtmf to answer |
ASTERISK-30423: Problem with segfault |
ASTERISK-30424: pjproject_bundled: cross-compilation broken when ssl autodetected |
ASTERISK-30425: cel: CEL APP_START and APP_END are incorrectly aggregated when same apps runs in sequence |
ASTERISK-30426: Potential typo in ast_writestream() |
ASTERISK-30428: bridging: Music on hold continues after INVITE with replaces |
ASTERISK-30429: res_sorcery_memory_cache: Memory leak when calling sorcery_memory_cache_open |
ASTERISK-30430: SEGFAULT ASTERISK |
ASTERISK-30431: res_config_pgsql: Allow connection via URL |
ASTERISK-30432: res_speech_aeap: Tight loop; high CPU usage |
ASTERISK-30433: http.c: Minor simplification to HTTP status output. |
ASTERISK-30434: Asterisk webrtc |
ASTERISK-30435: chan_iax2: IPv6 client can't qualify/register if 1st bind address is IPv4 and vice versa |
ASTERISK-30436: error when launching |
ASTERISK-30437: app_queue ability to start periodic announcements at a different time than the playing interval (frequency) |
ASTERISK-30438: app_osplookup: Remove obsolete sample config |
ASTERISK-30439: DTMF with direct media |
ASTERISK-30440: app_senddtmf: Add Flash AMI action |
ASTERISK-30441: func_json: Fix JSON parsing of complex objects |
ASTERISK-30442: make install-logrotate causes logrotate to fail on service restart |
ASTERISK-30443: app_queue: Add "Agent Abandoned" AMI event |
ASTERISK-30445: stasis: Off-nominal channel leave causes bridge to be destroyed |
ASTERISK-30446: bridge_builtin_features: add periodic beep option to one touch monitor |
ASTERISK-30447: Stasis/p:channel:all reaching 500 tasks |
ASTERISK-30448: using the new Spandsp FAX Driver 16.30.0 |
ASTERISK-30449: contrib: rc.archlinux.asterisk uses invalid redirect. |
ASTERISK-30450: res_crypto, res_pjsip: build failures on 32-bit architectures |
ASTERISK-30451: res_pjsip: Contact header set incorrectly for call redirect (302 Moved temp.) when external_* set |
ASTERISK-30452: Test Suite: Fix regression causing all builds to fail |
ASTERISK-30453: On external transfer Contact header rewriten with external_signaling_address |
ASTERISK-30454: res_pjsip_refer: NOTIFY sipfrag may terminate early depending on conditions |
ASTERISK-30455: Increase channel name column width on cli |
ASTERISK-30456: I am currently in the process of setting up Asterisk, but I am experiencing an issue where I cannot hear the voice of the caller. |
ASTERISK-30457: res_agi: RECORD FILE plays 2 beeps |
ASTERISK-30458: Display Name in From Header |
ASTERISK-30459: how to configure this to get a call on freebsd |
ASTERISK-30460: Broken community login |
ASTERISK-30461: Can't hear voice on calls |
ASTERISK-30462: res_musiconhold: Add looplast option |
ASTERISK-30463: asterisk AMI |
ASTERISK-30464: app_mixmonitor: Allow specifying which MixMonitor instance (or all of them) to mute/unmute using MixMonitorMute |
ASTERISK-30465: format_sln: add support for .slin files |
ASTERISK-30466: My php agi script is not working when i have changed php 7 to php 8. From other forum i see the same issue, they suggest to change {} to [], But i am unable to understand where i have to change. Please share phpagi.php for php8 or suggest |
ASTERISK-30467: channels Originate issue |
ASTERISK-30468: call file issues |
ASTERISK-30469: res_pjsip_pubsub: Regression for subscription shutdowns |
ASTERISK-30470: Cisco SPA3XX and SPA5XX not register with PJSIP TLS and LE certs |
ASTERISK-30472: pbx_ael: Literal usage for variables broken |
ASTERISK-30473: chan_pjsip, also return all codecs on a re-INVITE without SDP on a late offer |
ASTERISK-30474: res_prometheus provides broken description |
ASTERISK-30475: Asterisk producing ICE candidates with same priority |
ASTERISK-30476: Hangup Cause is not passed in ChannelHanguprequest |
ASTERISK-30477: res_srtp: ROC reset bugs |
ASTERISK-30478: asterisk critical reboot |
ASTERISK-30479: voicemail.conf: Comments about #include files are wrong |
ASTERISK-30480: dialplan reload warning refers to wrong priorities in dialplan |
ASTERISK-30481: chan_console: Hangs on placing calls |
ASTERISK-30482: AudioSocket: Lack of wait in loop causing high CPU usage |
ASTERISK-30483: logger: Allow filtering logs in CLI by channel |
ASTERISK-30485: res_pjsip_pubsub: Add APIs for new pubsub capabilities |
ASTERISK-30486: app_queue: Fix minor xmldoc issues |
ASTERISK-30487: Queue member pause state lost when reloading queues.conf |
ASTERISK-30488: say.c Time announcement does not say o'clock for the French language |
ASTERISK-30489: Asterisk crashed unexpected event I tried to upgrade to latest version |
ASTERISK-30490: stasis: Deleting bridge multiple times at same time may cause crash |
ASTERISK-30491: res_pjsip_session: Crash when incrementing session reference count |
ASTERISK-30492: app_queue periodic-announce-startdelay patch derefferences null qe.parent pointer. |
ASTERISK-30493: I don't know the exactly what happening and what causing this issue but the issue i am facing that"Getting Autodestruct dialog message and when it occurs it will stoping and drops the calls" |
ASTERISK-30497: ChannelRedirect aborting hangup handler execution |
ASTERISK-30498: Segfault / 16.22.0 |
ASTERISK-30499: inband_progress not honored with ivr/direct accept |
ASTERISK-30500: Caller name corruption in encodings other than UTF-8 |
ASTERISK-30501: Frack assertion error in while loop in stasis.c |
ASTERISK-30502: asterisk crashes when loading res_odbc |
ASTERISK-30503: how to link IVR to an api |
ASTERISK-30504: Support sending in-dialog messages to channel via ARI |